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Versions: (draft-shieh-rtcweb-ip-handling) 00 01 02 03 04

Network Working Group                                          J. Uberti
Internet-Draft                                                  G. Shieh
Intended status: Standards Track                                  Google
Expires: January 4, 2018                                    July 3, 2017


                WebRTC IP Address Handling Requirements
                    draft-ietf-rtcweb-ip-handling-04

Abstract

   This document provides information and requirements for how IP
   addresses should be handled by WebRTC implementations.

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
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   Internet-Drafts are draft documents valid for a maximum of six months
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   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on January 4, 2018.

Copyright Notice

   Copyright (c) 2017 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
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   (http://trustee.ietf.org/license-info) in effect on the date of
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   described in the Simplified BSD License.






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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   2
   3.  Problem Statement . . . . . . . . . . . . . . . . . . . . . .   2
   4.  Goals . . . . . . . . . . . . . . . . . . . . . . . . . . . .   4
   5.  Detailed Design . . . . . . . . . . . . . . . . . . . . . . .   4
   6.  Application Guidance  . . . . . . . . . . . . . . . . . . . .   6
   7.  Security Considerations . . . . . . . . . . . . . . . . . . .   6
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   6
   9.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .   7
   10. References  . . . . . . . . . . . . . . . . . . . . . . . . .   7
     10.1.  Normative References . . . . . . . . . . . . . . . . . .   7
     10.2.  Informative References . . . . . . . . . . . . . . . . .   7
   Appendix A.  Change log . . . . . . . . . . . . . . . . . . . . .   8
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .   9

1.  Introduction

   One of WebRTC's key features is its support of peer-to-peer
   connections.  However, when establishing such a connection, which
   involves connectivity tests using various IP addresses, WebRTC may
   allow a web application to learn additional information about the
   user compared to an application that only uses the Hypertext Transfer
   Protocol (HTTP) [RFC7230].  This may be problematic in certain cases.
   This document summarizes the concerns, and makes recommendations on
   how WebRTC implementations should best handle the tradeoff between
   privacy and media performance.

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [RFC2119].

3.  Problem Statement

   In order to establish a peer-to-peer connection, WebRTC
   implementations use Interactive Connectivity Establishment (ICE)
   [RFC5245], which gathers and exchanges all the IP addresses it can
   discover, using techniques like Session Traversal Utilities for NAT
   (STUN) [RFC5389] and Traversal Using Relays around NAT (TURN)
   [RFC5766], in order to check the connectivity of each local-address-
   remote-address pair and select the best one.  The addresses that are
   gathered usually consist of an endpoint's private physical/virtual
   addresses and its public Internet addresses.





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   These addresses are exposed upwards to the web application, so that
   they can be communicated to the remote endpoint.  This allows the
   application to learn more about the local network configuration than
   it would from a typical HTTP scenario, in which the web server would
   only see a single public Internet address, i.e. the address from
   which the HTTP request was sent.

   The information revealed falls into three categories:

   1.  If the client is behind a Network Address Translator (NAT), the
       client's private IP addresses, typically [RFC1918] addresses, can
       be learned.

   2.  If the client tries to hide its physical location through a
       Virtual Private Network (VPN), and the VPN and local OS support
       routing over multiple interfaces (i.e., a "split-tunnel" VPN),
       WebRTC will discover the public address for the VPN as well as
       the ISP public address that the VPN runs over.

   3.  If the client is behind a proxy (a client-configured "classical
       application proxy", as defined in [RFC1919], Section 3), but
       direct access to the Internet is also supported, WebRTC's STUN
       checks will bypass the proxy and reveal the public address of the
       client.

   Of these three concerns, #2 is the most significant concern, since
   for some users, the purpose of using a VPN is for anonymity.
   However, different VPN users will have different needs, and some VPN
   users (e.g.  corporate VPN users) may in fact prefer WebRTC to send
   media traffic directly, i.e., not through the VPN.

   #3 is a less common concern, as proxy administrators can control this
   behavior through organization firewall policy if desired, coupled
   with the fact that forcing WebRTC traffic through a proxy will have
   negative effects on both the proxy and on media quality.  For
   situations where this is an important consideration, use of a RETURN
   proxy, as described below, can be an effective solution.

   #1 is considered to be the least significant concern, given that the
   local address values often contain minimal information (e.g.
   192.168.0.2), or have built-in privacy protection (e.g.  [RFC4941]
   IPv6 addresses).

   Note also that these concerns predate WebRTC; Adobe Flash Player has
   provided similar functionality since the introduction of RTMFP
   [RFC7016] in 2008.





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4.  Goals

   Being peer-to-peer, WebRTC represents a privacy-enabling technology,
   and therefore we want to avoid solutions that disable WebRTC or make
   it harder to use.  This means that WebRTC should be configured by
   default to only reveal the minimum amount of information needed to
   establish a performant WebRTC session, while providing options to
   reveal additional information upon user consent, or further limit
   this information if the user has specifically requested this.
   Specifically, WebRTC should:

   o  Provide a privacy-friendly default behavior which strikes the
      right balance between privacy and media performance for most users
      and use cases.

   o  For users who care more about one versus the other, provide a
      means to customize the experience.

5.  Detailed Design

   The key principles for the design are listed below:

   1.  By default, WebRTC should follow normal IP routing rules, to the
       extent that this is easy to determine (i.e., not considering
       proxies).  This can be accomplished by binding local sockets to
       the wildcard addresses (0.0.0.0 for IPv4, :: for IPv6), which
       allows the OS to route WebRTC traffic the same way as it would
       HTTP traffic, and allows only the 'typical' public addresses to
       be discovered.

   2.  By default, support for direct connections between hosts (i.e.,
       without traversing a NAT or relay server) should be maintained.
       To accomplish this, the local IPv4 and IPv6 addresses of the
       interface used for outgoing STUN traffic should still be surfaced
       as candidates, even when binding to the wildcard addresses as
       mentioned above.  The appropriate addresses here can be
       discovered by the common trick of binding sockets to the wildcard
       addresses, connect()ing those sockets to some well-known public
       IP address, and then reading the bound local addresses via
       getsockname().  This approach requires no data exchange; it
       simply provides a mechanism for applications to retrieve the
       desired information from the kernel routing table.

   3.  Determining whether a web proxy is in use is a complex process,
       as the answer can depend on the exact site or address being
       contacted.  Furthermore, web proxies that support UDP are not
       widely deployed today.  As a result, when WebRTC is made to go
       through a proxy, it typically needs to use TCP, either ICE-TCP



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       [RFC6544] or TURN-over-TCP [RFC5766].  Naturally, this has
       attendant costs on media quality as well as proxy performance,
       and should be avoided where possible.

   4.  RETURN [I-D.ietf-rtcweb-return] is a proposal for explicit
       proxying of WebRTC media traffic.  When RETURN proxies are
       deployed, media and STUN checks will go through the proxy, but
       without the performance issues associated with sending through a
       typical web proxy.

   Based on these ideas, we define four specific modes of WebRTC
   behavior, reflecting different media quality/privacy tradeoffs:

   Mode 1:  Enumerate all addresses: WebRTC MUST bind to all interfaces
            individually and use them all to attempt communication with
            STUN servers, TURN servers, or peers.  This will converge on
            the best media path, and is ideal when media performance is
            the highest priority, but it discloses the most information.

   Mode 2:  Default route + associated local addresses: WebRTC MUST
            follow the kernel routing table rules (e.g., by binding
            solely to the wildcard address), which will typically cause
            media packets to take the same route as the application's
            HTTP traffic.  In addition, any private IPv4 and IPv6
            addresses associated with the kernel-chosen interface MUST
            be discovered through getsockname, as mentioned above, and
            provided to the application.  This ensures that direct
            connections can still be established in this mode.

   Mode 3:  Default route only: This is the the same as Mode 2, except
            that the associated private address MUST NOT be provided.
            This may cause traffic to hairpin through a NAT, fall back
            to the application TURN server, or fail altogether, with
            resulting quality implications.

   Mode 4:  Force proxy: This forces all WebRTC media traffic through a
            proxy, if one is configured.  If the proxy does not support
            UDP (as is the case for all HTTP and most SOCKS [RFC1928]
            proxies), or the WebRTC implementation does not support UDP
            proxying, the use of UDP will be disabled, and TCP will be
            used to send and receive media through the proxy.  Use of
            TCP will result in reduced quality, in addition to any
            performance considerations associated with sending all
            WebRTC media through the proxy server.

   Mode 1 MUST only be used when user consent has been provided; this
   thwarts the typical drive-by enumeration attacks.  The details of




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   this consent are left to the implementation; one potential mechanism
   is to tie this consent to getUserMedia consent.

   In cases where user consent has not been obtained, Mode 2 SHOULD be
   used.  This allows applications to still achieve direct connections
   in many cases, even without consent (e.g., data channel
   applications).  However, user agents MAY choose a stricter default
   policy in certain circumstances.

   Note that when a RETURN proxy is configured for the interface
   associated with the default route, Mode 2 and 3 will cause any
   external media traffic to go through the RETURN proxy.  While the
   RETURN approach gives the best performance, a similar result can be
   achieved for non-RETURN proxies via an organization firewall policy
   that only allows external WebRTC traffic to leave through the proxy
   (typically, over TCP).  This provides a way to ensure the proxy is
   used for any external traffic, but avoids the performance issues of
   Mode 4, where all media is forced through said proxy, for intra-
   organization traffic.

6.  Application Guidance

   The recommendations mentioned in this document may cause certain
   WebRTC applications to malfunction.  In order to be robust in all
   scenarios, the following guidelines are provided for applications:

   o  Applications SHOULD deploy a TURN server with support for both UDP
      and TCP connections to the server.  This ensures that connectivity
      can still be established, even when Mode 3 or 4 are in use,
      assuming the TURN server can be reached.

   o  Applications SHOULD detect when they don't have access to the full
      set of ICE candidates by checking for the presence of host
      candidates.  If no host candidates are present, Mode 3 or 4 above
      is in use; this knowledge can be useful for diagnostic purposes.

7.  Security Considerations

   This document is entirely devoted to security considerations.

8.  IANA Considerations

   This document requires no actions from IANA.








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9.  Acknowledgements

   Several people provided input into this document, including Bernard
   Aboba, Harald Alvestrand, Ted Hardie, Matthew Kaufmann, Eric
   Rescorla, Adam Roach, and Martin Thomson.

10.  References

10.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,
              <http://www.rfc-editor.org/info/rfc2119>.

10.2.  Informative References

   [I-D.ietf-rtcweb-return]
              Schwartz, B. and J. Uberti, "Recursively Encapsulated TURN
              (RETURN) for Connectivity and Privacy in WebRTC", draft-
              ietf-rtcweb-return-02 (work in progress), March 2017.

   [RFC1918]  Rekhter, Y., Moskowitz, B., Karrenberg, D., de Groot, G.,
              and E. Lear, "Address Allocation for Private Internets",
              BCP 5, RFC 1918, DOI 10.17487/RFC1918, February 1996,
              <http://www.rfc-editor.org/info/rfc1918>.

   [RFC1919]  Chatel, M., "Classical versus Transparent IP Proxies",
              RFC 1919, DOI 10.17487/RFC1919, March 1996,
              <http://www.rfc-editor.org/info/rfc1919>.

   [RFC1928]  Leech, M., Ganis, M., Lee, Y., Kuris, R., Koblas, D., and
              L. Jones, "SOCKS Protocol Version 5", RFC 1928,
              DOI 10.17487/RFC1928, March 1996,
              <http://www.rfc-editor.org/info/rfc1928>.

   [RFC4941]  Narten, T., Draves, R., and S. Krishnan, "Privacy
              Extensions for Stateless Address Autoconfiguration in
              IPv6", RFC 4941, DOI 10.17487/RFC4941, September 2007,
              <http://www.rfc-editor.org/info/rfc4941>.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              DOI 10.17487/RFC5245, April 2010,
              <http://www.rfc-editor.org/info/rfc5245>.





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   [RFC5389]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
              "Session Traversal Utilities for NAT (STUN)", RFC 5389,
              DOI 10.17487/RFC5389, October 2008,
              <http://www.rfc-editor.org/info/rfc5389>.

   [RFC5766]  Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
              Relays around NAT (TURN): Relay Extensions to Session
              Traversal Utilities for NAT (STUN)", RFC 5766,
              DOI 10.17487/RFC5766, April 2010,
              <http://www.rfc-editor.org/info/rfc5766>.

   [RFC6544]  Rosenberg, J., Keranen, A., Lowekamp, B., and A. Roach,
              "TCP Candidates with Interactive Connectivity
              Establishment (ICE)", RFC 6544, DOI 10.17487/RFC6544,
              March 2012, <http://www.rfc-editor.org/info/rfc6544>.

   [RFC7016]  Thornburgh, M., "Adobe's Secure Real-Time Media Flow
              Protocol", RFC 7016, DOI 10.17487/RFC7016, November 2013,
              <http://www.rfc-editor.org/info/rfc7016>.

   [RFC7230]  Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
              Protocol (HTTP/1.1): Message Syntax and Routing",
              RFC 7230, DOI 10.17487/RFC7230, June 2014,
              <http://www.rfc-editor.org/info/rfc7230>.

Appendix A.  Change log

   Changes in draft -04:

   o  Rewording and cleanup in abstract, intro, and problem statement.

   o  Added 2119 boilerplate.

   o  Fixed weird reference spacing.

   o  Expanded acronyms on first use.

   o  Removed 8.8.8.8 mention.

   o  Removed mention of future browser considerations.

   Changes in draft -03:

   o  Clarified when to use which modes.

   o  Added 2119 qualifiers to make normative statements.

   o  Defined 'proxy'.



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   o  Mentioned split tunnels in problem statement.

   Changes in draft -02:

   o  Recommendations -> Requirements

   o  Updated text regarding consent.

   Changes in draft -01:

   o  Incorporated feedback from Adam Roach; changes to discussion of
      cam/mic permission, as well as use of proxies, and various
      editorial changes.

   o  Added several more references.

   Changes in draft -00:

   o  Published as WG draft.

Authors' Addresses

   Justin Uberti
   Google
   747 6th St S
   Kirkland, WA  98033
   USA

   Email: justin@uberti.name


   Guo-wei Shieh
   Google
   747 6th St S
   Kirkland, WA  98033
   USA

   Email: guoweis@google.com













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